What CleanCutAudio is and who this manual is for.
CleanCutAudio is an audio editor for Windows designed to clean up, polish and “master” recordings: voice, podcasts, music, samples. It works with MP3 and WAV files and offers three groups of tools: a waveform editor (cuts, fades, silence, reverse, pitch), a professional DSP effects chain (noise reduction, de-esser, equalizer, compressor, ambient, soft clipper, gain, stereo widener) and a set of analysis tools (LUFS/loudness, FFT spectrum analyzer, VU meters, mastering report).
This manual is meant to be read on two levels. The main text explains what each feature does and how to use it, in plain language. The blue-gray boxes marked ⚙️ Technical deep dive add details on the parameters (thresholds, ratio, LUFS, true peak…) for those who want to understand what happens “under the hood”. If you are a beginner you can safely skip the technical boxes: the rest of the text is all you need to get to work.
Cuts, regions, fades and silence applied on the waveform with draggable markers.
Eight modules in series, with ready-made presets and real-time preview while you listen.
LUFS loudness to EBU R128, true peak, FFT spectrum and mastering report.
MP3 (CBR/VBR) and WAV, per region or whole file, with ID3 tags.
How the main window is organized (v2.2.0.0).
At startup, until you load a file, the waveform shows a welcome screen inviting you to drag in an MP3 or WAV, plus a reminder of the main shortcuts. The window is divided into a few fixed areas:
| Area | What it does |
|---|---|
| Menu bar | At the top: File · Project · Edit · Cuts&Regions · Processing · Tools · Voice&Text · Audio Device · Settings · Help (see the table below). |
| File browser | Tree panel with your computer's folders, favorites and quick locations (C:\, Music, Desktop, Documents), a Change Directory button and a Text Search (F3 = Next) box. Double-click a file or press Enter to load it. |
| Information panel | Top left: MP3 Loaded Data (file name, folder, duration, sample rate, channels, bit depth, total samples, bitrate, size) and ID3 Data (title, artist, year). |
| Waveform | The central area where you view and edit the audio: left channel (blue) and right channel (red), markers, regions, fades, playhead (cyan line), time ruler. The portions outside the selection are dimmed and marked REMOVED SECTION. |
| Quick buttons (right) | DSP, Normalize LUFS, Sound Analyzer, STT Transcribe, STT Transcript, TTS TextToSpeech: shortcuts to the main windows and tools. |
| VU meters | Two radial Left / Right meters on the right, showing the level in real time. |
| Transport (bottom) | Play (green), Stop (red), Record (recording), PlayHead readout with time and percentage, Enable Loop and Follow PlayHead checkboxes. At the very bottom, the status bar shows the file path and the active audio driver (e.g. WASAPI (…) – Shared). |
| Menu | Main items |
|---|---|
| File | 📂 Open… · 💾 Save As… · ✏️ Edit mp3 Info… |
| Project | 📦 Open Project… · 📥 Save Project… · 💾 Save Project As Single File… |
| Edit | Insert Silence · Reverse · Cut/Copy/Paste · Paste File At Playhead · Pitch Shift… · Fix DC Offset · Trim Start/End Silence · Trim Internal Silence · Auto De-Click · Trim to Selection · Trim to Active Region · Undo |
| Cuts&Regions | Insert Marker Cut · Delete Marker Cut · Remove All Markers Cut · Trim to Active Region |
| Processing | RealTimeDSP… (the DSP chain) · Sound Analyzer… |
| Tools | Loudness Normalization (LUFS)… · Recorder… · Tone Generator… |
| Voice&Text | Text-to-Speech Module ▸ System Voice… · Speech-to-Text Module ▸ Convert Speech to Text / View Transcript Form… |
| Audio Device | Choice of audio driver and device. |
| Settings | Export Settings… · DSP Settings… |
| Help | About… |
Many features also have a quick button in the right-hand column or a keyboard shortcut. The DSP window (effects chain) and the Sound Analyzer open separately and can stay open while you work.
Three ways to bring audio into the editor.
In the browser you can add the folders you use often to your Favorites (right-click ▸ Add to Favorites) and jump quickly to standard locations with the roots button. Once loaded, the file appears in the waveform and its data shows up in the information panel.
The heart of editing: marking out the portions you want to work on.
CleanCutAudio distinguishes two concepts that are best kept separate:
Drag the two markers directly on the waveform to define the selection. During playback the selection stays consistent: if you move the markers, the playback position is brought back inside the range.
Click a region to select it: it becomes the “active” region, the one the DSP window and several commands refer to. Press F10 to show/hide the list of regions and cuts.
DspConfig). This means you can apply, for example, heavy noise reduction only to the spoken intro and different compression to the chorus, all in the same file. On export, each region is rendered with its own chain. The central concept of CleanCutAudio — where regions come from and everything that follows from them.
The cut marker (marker cut) is the element the program's entire workflow revolves around. It is not just a decorative “notch”: each marker cut divides the audio into regions and each region becomes a self-contained block, with its own fades, its own DSP chain and its own export behavior. Understanding marker cuts well means understanding how CleanCutAudio thinks.
When you insert a marker cut, the audio to the left and to the right of the cut becomes two separate regions. Adding more gives you a sequence of consecutive regions covering the whole file. Each region carries:
Each region has a name and a color, so you can recognize it at a glance on the waveform and in the lists.
Noise reduction, EQ, compressor… all modules have independent settings for each region.
Fade-in and fade-out are defined per region, with dedicated handles.
On export, each region is rendered with its own chain and its own range.
| Operation | How to do it | Result |
|---|---|---|
| Insert a cut | + at the desired point, or Cuts/Regions ▸ Insert Marker Cut | The audio is split into two regions at the cut point. |
| Move a cut | Drag the marker cut on the waveform | The adjacent regions resize accordingly, in real time. |
| Remove a single cut | −, or Cuts/Regions ▸ Delete Marker Cut | The two regions on either side merge back into one. |
| Remove all cuts | Ctrl+Shift+X, or Remove All Markers Cut | You are back to a single region (the main region). |
| Show/hide the TIMELINE overlay | F10 | Shows the chronological list of markers, regions and cuts with their exact times. |
Pressing F10 brings up the TIMELINE overlay, the chronological map of the file: it lists, in order, MARKER START, each REGION with its range (e.g. Region 03 [1:46.237 → 2:27.690]), each CUT with its position and finally MARKER END. Regions are named automatically (Region 01, 02, …) and cuts stay “(unnamed)” until you rename them; the entry for the selected region is highlighted. It is the most accurate way to check cut points down to the millisecond.
Clicking a region makes it active: it is the region the DSP window (which shows the region's name and color at the top) and several editing commands refer to. In other words, to apply effects or actions to a specific portion, first you select it, then you act. Hovering the mouse over regions highlights them without activating them.
This is the most powerful consequence: since each region can have an independent DSP chain, you can treat different parts of the same file differently. Typical examples: heavy noise cleanup only on the spoken intro; more compression on the chorus; brighter highs only on a poorly recorded verse; turning the de-esser off where it is not needed.
Each region has its own fade-in/fade-out handles. When you drag the marker cut between two regions, the program prevents the fades of neighboring regions from overlapping beyond the allowed limit, so the join point stays free of clicks and gaps. It is the correct way to join two tracks or split a medley.
With Cuts/Regions ▸ Trim to Active Region (or Edit ▸ Trim to Active Region) you discard everything else and keep only the selected region: useful for extracting a single song from a long recording split with markers.
On export, the summary lists each region with its range and its DSP chain. With no cuts, the single main region is exported (the range between the start/end markers); with several cuts, each region is rendered with its own settings.
AudioRegion) contains its own boundary samples (StartSample/EndSample), fade durations in seconds, color, name and a complete DSP configuration (DspConfig). Moving a marker cut updates the boundaries of the adjacent regions in real time, with a check that blocks the move if the fades would overlap (fadeIn + margin ≥ fadeOut). Marker cuts have hit-test priority over regions: when the pointer is within a few pixels of a cut, the click grabs the cut and selects its associated region. Smooth fade-ins and fade-outs, with no clicks.
Each region (and the main region when there are no cuts) has two fade handles: one at the top left for the fade-in and one at the top right for the fade-out. Drag them toward the center to lengthen the fade, toward the edges to shorten it. The fade-in and fade-out durations are stored for each region.
Fades are used to avoid “clicks” at the ends of a cut, to blend two adjacent regions and to end a track gracefully. When you drag a cut marker between two regions, the program prevents the fades of neighboring regions from overlapping beyond the allowed limit, so the transition stays clean.
Listening to your work, with or without effects.
All the operations that change the audio, from the Edit menu.
These commands act on the selection or on the active region, depending on the case. You can always go back with Undo.
| Command | Shortcut | What it does |
|---|---|---|
| Insert Silence | Ctrl+W | Zeroes out (silences) the selected portion while keeping its duration. Useful for removing a noise or a word without shifting the rest. |
| Reverse | Ctrl+R | Reverses the portion (makes it play backwards). |
| Cut / Copy / Paste | — / — / Ctrl+P | Cuts, copies and pastes portions of audio via the internal clipboard. |
| Paste File At Playhead | — | Inserts the contents of another audio file at the playhead position. |
| Fix DC Offset | — | Removes the DC component that shifts the waveform above or below zero (a cause of clicks and lost headroom). |
| Trim Start/End Silence | — | Removes the silence at the beginning and end of the file. |
| Trim Internal Silence | — | Shortens or removes overly long pauses within the audio (pauses between sentences, gaps). |
| Auto De-Click | — | Automatically attenuates clicks and impulsive noise (typical of vinyl or splices). |
| Trim to Selection | — | Discards everything outside the selection, keeping only the range between the two markers. |
| Trim to Active Region | — | Keeps only the currently selected region. |
| Undo | — | Undoes the last operation. |
Raising or lowering the pitch (Edit ▸ Pitch Shift…).
Pitch Shift changes the pitch of the current region/selection. The window offers a slider in semitones, a Preview button (listen to a preview of about 2 seconds before applying) and the Apply and Cancel buttons.
The processing modules in series, with real-time preview.
Open the window with the DSP button, from the Processing ▸ RealTimeDSP… menu or with the F9 key. The DSP chain is the set of effects the audio passes through in sequence: the output of one module is the input of the next. Each region has its own chain, so the window always shows the parameters of the active region — the current region is shown at the top (e.g. Region: Main Region (MASTER)).
The recommended order (“PRO order”), from top to bottom, is the order in which the signal is processed:
| # | Module | Its role in a word |
|---|---|---|
| 1 | Noise Reduction | Background hiss cleanup |
| 2 | De-Esser | Sibilance control |
| 3 | Equalizer | Tonal balance |
| 4 | Compressor | Dynamics control |
| — | Ambient | Sense of space/air |
| 5 | SoftClipper | Saturation and soft limiting |
| 6 | Gain | Final volume |
At the top of the list you will also find the “fixed” modules Stereo Widener and Stereo Linking (L/R channel linking), which act on the overall stereo image.
In the pages that follow, each module has a card describing what it is for, how to set it up, a table of its parameters with their ranges, and the factory presets.
Reduces constant background noise — hiss, hum, microphone noise — by tracking the noise level and lowering the signal when it drops below that threshold. It is the first link in the chain because it is best to clean up before equalizing and compressing.
Start from a preset (Light / Medium / Strong Clean) and loop a part with only noise and a part with the voice. Raise the Sensitivity until the background disappears, but stop as soon as the voice starts to sound “muffled” or to open and close unnaturally. The Reduction sets how deep the cut goes.
| Parameter | Range | Effect |
|---|---|---|
| Sensitivity | 0 – 100 | How aggressively the noise is tracked: higher = kicks in sooner. |
| Reduction | 0 – 30 dB | Maximum attenuation applied below the threshold. |
| Attack | ms | How quickly the reduction kicks in when the signal drops. |
| Release | ms | How quickly the reduction releases when the signal returns. |
| Preset | Sensitivity | Reduction | Attack | Release |
|---|---|---|---|---|
| Light Clean | 35 | 8 dB | 15 ms | 180 ms |
| Medium Clean | 50 | 12 dB | 20 ms | 220 ms |
| Strong Clean | 65 | 18 dB | 25 ms | 260 ms |
Softens sibilance — the “s”, “sh” and “z” sounds that become harsh and piercing, especially on close-miked voice. It acts only on the offending frequency band and only when that band exceeds a threshold.
| Parameter | Range | Effect |
|---|---|---|
| Threshold | −60 … 0 dB | Threshold above which the sibilant is attenuated. |
| Ratio | 1 … 10 | Amount of reduction (higher = more strongly damped sibilant). |
| Frequency | 3,000 … 12,000 Hz | Center of the sibilance band. |
| Bandwidth | 500 … 6,000 Hz | Width of the controlled band. |
| Solo Sibilants | on/off | Listen to the band in isolation, to center it. |
| Preset | Threshold | Ratio | Frequency | Bandwidth |
|---|---|---|---|---|
| SoftVoice | −25 dB | 2.5 | 5,500 Hz | 2,500 Hz |
| FemaleVocal | −28 dB | 3.5 | 6,500 Hz | 3,000 Hz |
| MaleVocal | −24 dB | 3.0 | 5,000 Hz | 2,500 Hz |
| PodcastClean | −22 dB | 2.0 | 6,000 Hz | 2,000 Hz |
| StrongSibilance | −30 dB | 5.0 | 7,000 Hz | 3,500 Hz |
Frequency / Bandwidth and limited between 0.3 and 12. When the energy in the band exceeds the threshold, a reduction proportional to the ratio is applied, with gain smoothing so as not to create artifacts. It is a band-detection de-esser, typical of vocal plugins. Corrects the tonal balance: removes boominess from the lows, adds presence and “air” to the highs, carves out annoying frequencies. It is a 5-band parametric equalizer, with each band configurable independently.
For each band, choose the filter type, the frequency, the gain (cut or boost) and the Q (width of the adjustment). The frequency is adjusted on a logarithmic scale, as in professional plugins, so you get fine resolution in the lows.
| Parameter | Description |
|---|---|
| Type | Filter type: Peaking (bell), HighPass (cuts the lows below the frequency), LowPass (cuts the highs above the frequency) and others. |
| Frequency | Center/cutoff frequency of the band (logarithmic scale). |
| Gain (dB) | How much to boost (+) or attenuate (−) around that frequency. |
| Q | Width of the adjustment: high Q = narrow, surgical adjustment; low Q = broad, musical curve. |
The included presets are: LowEnd Cut (cleans up excess lows), Bass Boost, Treble Boost, Air Boost (opens up the highs), Voice Clean, Podcast Voice (low cut + presence on speech), Telephone Effect (“telephone” voice), HiFi Smile Curve (boosted lows and highs) and Mastering Gentle Curve (a gentle mastering smile curve). Each preset sets the 5 bands to ready-made values that you can then fine-tune.
freq = fMin·(fMax/fMin)^knob), while gain and Q are linear. The presets define frequency, gain, Q and type for each of the 5 bands — for example “Podcast Voice” combines a HighPass at 80 Hz, a cut at 150 Hz, a presence boost at 2.5 kHz and a safety LowPass. Evens out the dynamics: it reduces the gap between loud and soft parts, making the voice more consistent and present and the mix more “compact”. It is the module with the most parameters, with control over the knee, makeup and output ceiling.
| Parameter | Range | Effect |
|---|---|---|
| Threshold | −60 … 0 dB | Level above which compression starts. |
| Ratio | 1 … 30 | Compression ratio (4 = 4:1). Higher = firmer. |
| Knee | 0 … 1 | “Knee”: 0 = abrupt action (hard knee), 1 = gradual onset (soft knee). |
| Attack | 0.1 … 200 ms | How quickly it reacts to peaks. |
| Release | 5 … 2000 ms | How quickly it releases after the peak. |
| Makeup | −12 … +24 dB | Makeup gain after compression. |
| Ceiling | −12 … 0 dB | Maximum output ceiling. |
| Preset | Threshold | Ratio | Makeup | Typical use |
|---|---|---|---|---|
| SoftVocal | −24 dB | 2.2 | +3.5 dB | Delicate voice, light compression |
| Podcast | −20 dB | 3.2 | +3.5 dB | Steady, present speech |
| MusicPunch | −18 dB | 3.8 | +2.5 dB | Music with more “punch” |
| HardLimiter | −6 dB | 20 | 0 dB | Firm peak limiting |
| MasteringGlue | −12 dB | 2.0 | +2 dB | Gentle mastering “glue” |
tanh (LUT). With CompLink linking enabled, Left and Right share the same parameters. Adds a sense of space and air around the sound, like a small room/short reverb. Used sparingly it gives depth to the voice; pushed harder, it creates a “dreamy” effect.
| Parameter | Effect |
|---|---|
| Mix | Amount of effect blended into the dry sound (0 = none). |
| Delay | Delay that defines the perceived “size” of the space (ms). |
| Preset | Mix | Delay |
|---|---|---|
| Soft | 10% | 5 ms |
| Warm | 18% | 7.5 ms |
| Wide Room | 25% | 10 ms |
| Airy | 35% | 12.5 ms |
| Dreamy | 55% | 20 ms |
Adds warmth and “glue”, and at the same time acts as a soft limit on peaks: instead of clipping abruptly (hard clip), it rounds off the crests, gaining perceived loudness without unpleasant distortion.
| Parameter | Effect |
|---|---|
| Drive | How hard to “push” the signal into saturation (more drive = more warmth/density). |
| Output | Output level, to compensate for the increase in volume. |
| Ceiling | Maximum ceiling of the output signal. |
| Preset | Drive | Output | Ceiling |
|---|---|---|---|
| Gentle | 2 dB | −1 dB | −1 dB |
| Warm | 4 dB | −2 dB | −1 dB |
| Punch | 6 dB | −3 dB | −1 dB |
| Hard | 9 dB | −4 dB | −1 dB |
The last link: it sets the final volume of the region and can apply a fade-in and fade-out at module level. This is where you set the final level before export or LUFS normalization.
| Parameter | Effect |
|---|---|
| Gain (dB) | Overall gain or attenuation. |
| Fade In / Fade Out | Fade-in/fade-out expressed in samples. |
| Preset | Gain | Fade In | Fade Out |
|---|---|---|---|
| Unity | 0 dB | 0 | 0 |
| Boost Smooth | +6 dB | 2000 | 2000 |
| Cut Smooth | −6 dB | 2000 | 2000 |
| Boost Hard | +12 dB | 500 | 500 |
| Cut Hard | −12 dB | 500 | 500 |
The Stereo Widener widens the stereo image, giving the sound a sense of spaciousness and “width”. Stereo Linking is the switch that links the Left and Right channels of the modules, so a change on one side applies to both.
| Parameter (Widener) | Effect |
|---|---|
| Delay | Inter-channel delay that creates the widening (ms). |
| Mix | Amount of effect applied. |
| Preset | Delay | Mix |
|---|---|---|
| Soft | 5 ms | 100% |
| Medium | 10 ms | 100% |
| Hard | 40 ms | 100% |
Measuring and normalizing perceived loudness to the EBU R128 standard.
LUFS (Loudness Units Full Scale) measures how loud audio is to the human ear, not just technically. It is the yardstick streaming platforms use to match volumes: if your track is too loud, Spotify or YouTube turn it down; if it is too quiet, it falls behind the others. The LUFS Normalization window analyzes the file and tells you exactly where you are and how much gain you need to hit a target.
Open it from the Normalize LUFS button or from Tools ▸ Loudness Normalization (LUFS)….
| Command | What it does |
|---|---|
| Analyze LUFS | Measures the track and fills in the data and the graph. |
| Preview LUFS | Plays a preview with the normalization gain applied. |
| Apply Normalization | Applies the gain to the track. |
| Undo Normalization | Undoes the applied normalization. |
| Preset / Target LUFS / Ceiling (dBTP) | Choose the target preset or set the target and ceiling by hand. |
| Preset | Target | Ceiling | True Peak Limiter |
|---|---|---|---|
| StreamingStandard | −14 LUFS | −1.0 dB | ON |
| YouTube | −14 LUFS | −1.5 dB | ON |
| Spotify | −14 LUFS | −2.0 dB | ON |
| AppleMusic | −16 LUFS | −1.0 dB | ON |
| BroadcastEBU | −23 LUFS | −1.0 dB | ON |
| LoudnessWar | −9 LUFS | 0 dB | OFF |
| NoLimiter | −14 LUFS | 0 dB | OFF |
| TruePeakOnly | 0 LUFS | −1.0 dB | ON |
| Measurement | Meaning | Reference values |
|---|---|---|
| Integrated LUFS | Average loudness of the whole track (the main EBU R128 measurement). | Streaming ≈ −14 · Podcast ≈ −16 · Broadcast TV −23 · Cinema −27 |
| Short-Term LUFS | Loudness over 3 s windows (trend section by section). Min = quietest part, Max = loudest. | — |
| Momentary LUFS | Loudness over 400 ms windows (very responsive). | — |
| LRA (Loudness Range) | Perceived dynamics = Short-Term Max − Min. | Compressed pop 2–4 · Rock 4–8 · Jazz 8–14 · Classical 14+ |
| RMS | Average energy (technical reference). | Music −9/−12 dBFS · Speech −15/−20 dBFS |
| Peak (dBFS) | Maximum digital sample peak. | Correct master ≤ −1.0 dBFS |
| True Peak (dBTP) | Actual peak after analog reconstruction (oversampling): the real risk of clipping on DACs and streaming. | Streaming ≤ −1.0 dBTP |
| PLR (Peak-Loudness Ratio) | Peak − LUFS: an index of dynamic “breathing room”. Low = heavily compressed. | < 8 critical · 8–12 caution · > 12 good |
| Gain to Target | Required gain = Target − Integrated. | e.g. −18 → target −14 = +4 dB |
At the bottom of the window, the NORMALIZE MULTIPLE FILES (LUFS) module normalizes several files at once with the same preset: add the files with Add Files… (or empty the list with Clear List), choose the output folder with Browse, then Start Batch (or Stop Batch). A Log and progress bars track the process. Handy for evening out the volume of an entire collection.
min(GainToTarget, GainAllowed), so raising the volume never exceeds the ceiling you set. When the preset has the True Peak Limiter enabled, a limiter with 4× oversampling (typical threshold −1.0 dBTP, with knee and lookahead) holds back the true peaks that would exceed the ceiling, avoiding distortion on DACs and streaming platforms. Measuring the quality of your track and comparing it with a reference track.
Open the Sound Analyzer from the button of the same name or from Processing ▸ Sound Analyzer…. It is not (just) a spectrum: it is a measurement station that, when you press Process And Analyze, calculates a set of indicators on the selected region and lets you compare them with a reference track that sounds “the way you want”.
| Abbreviation | Name | What it indicates |
|---|---|---|
| PLR | Peak-Loudness Ratio | How compressed/dynamic the track is: high PLR = more “breathing room”, low = more compressed. |
| RMS | Average energy | Average loudness and density: high RMS = denser mix, low = more open and clean. |
| CF | Crest Factor | Strength of transients relative to the average level: high = lively transients, low = flattened. |
| TR | Transient Ratio | Presence and clarity of attacks: high = strong impact, low = soft/squashed attacks. |
| DR | Dynamic Range | How much the track “breathes” (overall dynamics). |
| LUFS | Integrated LUFS | Average perceived loudness (see the dedicated chapter). |
| MMC | Low-Level Energy (LLE) | Minimum level of the musical content: the useful “floor” of the signal. |
Below the meters, the Loaded File Analysis panel breaks the track down into bands — Sub, Bass, LowMid, Mid, HighMid, High — and reports Fast LUFS, PLR, Low-Level Energy and Peak Sample. The Reference panel next to it shows the same values for the reference track, so you can see at a glance where your mix has too much or too little.
| Command | What it does |
|---|---|
| Process And Analyze | Runs the analysis on the current region and updates the meters and bands. |
| Save As Reference | Saves the current analysis as a .ccaref reference file. |
| Load Reference | Loads a saved reference into the Reference panel. |
| COMPARE | Compares the loaded track and the reference, showing the deviations (Δ) band by band. |
| Generate Report | Produces the complete Mastering Report of the comparison. |
The actual spectrum analyzer (20 Hz → 20 kHz, with a frequency/level tooltip) is built into the top of the DSP window, together with the stereo vectorscope. Its typical parameters:
| Parameter | What it controls | Notes |
|---|---|---|
| FFT Size | Samples used for the transform (512 → 8192). | Larger = more frequency resolution but slower updates. |
| Window | Window: Hann, Hamming, Blackman, Rectangular. | Hann is the best compromise; Rectangular = more leakage. |
| Frequency Resolution | Spacing between the “bins”. | = SampleRate / FFTSize (e.g. 44100/4096 ≈ 10.7 Hz). |
| Magnitude (dBFS) | Amplitude in decibels; 0 dBFS = digital maximum. | Typical: −20 / −120 dBFS. |
| Peak Hold | Holds the maximum of each bin. | Off · Short · Long · Infinite. |
| Averaging | Time averaging of the spectrum. | None (responsive) · Fast · Slow (stable). |
| Frequency scale | Logarithmic or linear. | Log = like the ear, standard in pro plugins. |
| Smoothing | Smooths the spectrum (1/3, 1/6, 1/12 octave). | Useful for music analysis. |
A text report that sums up the analysis of the track.
The Mastering Report gathers the technical information and the suggestions about the current track in a single screen, organized into sections:
| Section | Contents |
|---|---|
| Main Info | File name, technical data, region info and DSP status. |
| Notes | Readable mastering notes (comparison with the references). |
| Engineer Mode | Detailed “sound engineer” technical notes. |
| AI Chain | Suggested processing chain. |
| Text Report | Complete text report with the comparisons. |
| Dynamics Curve | Dynamics trend + dedicated notes. |
The report's Export button copies the entire text to the clipboard, ready to paste into a document or an email.
Saving the result with the DSP chain applied.
Exporting produces the final file with all the processing applied (each region's DSP, fades, any normalization). Before you proceed, the summary window (Export Summary) shows what will be exported: for each region its range and active DSP chain, or — if there are no cuts — the whole main region.
| Setting | Options |
|---|---|
| Format | MP3 or WAV |
| MP3 mode | CBR (constant bitrate) or VBR (variable) |
| MP3 bitrate | from 96 to 320 kbps |
| Stereo Widener | You can include or exclude it in the exported file |
DspConfig and concatenated; with no cuts, the range of the main region (the start/end markers) is exported. The output formats are MP3 (LAME encoder, CBR at the chosen bitrate or a VBR preset) and WAV (16-bit PCM): the save dialog offers one or the other based on the default format. The format, the CBR/VBR mode and the bitrate are read from the preferences (DefaultExportFormat, DefaultMp3Mode, DefaultMp3Bitrate) and can be changed in the settings. ID3 tags are written to MP3 files only. The MP3 metadata: title, artist, album, cover art.
From File ▸ ✏️ Edit mp3 Info… (or from the export summary) you open the ID3 tag editor. Here you fill in the information that music players display: title, artist, album, year, genre and — depending on the build — the cover art. Filling in the tags is especially useful for podcasts and collections, where files are organized and displayed based on this data.
Saving your work to pick it up later.
A project stores not only the audio but also your editing work: markers, regions, fades and the DSP chains of each region. That way you can close and pick up exactly where you left off.
| Command (Project menu) | What it does |
|---|---|
| 📦 Open Project… | Opens a saved project. |
| 📥 Save Project… | Saves the current working state. |
| 💾 Save Project As Single File… | Saves everything (audio + settings) in a single self-contained file, handy for archiving or moving. |
Capturing the audio coming out of your computer (Tools ▸ Recorder…).
The recorder captures system audio — what you hear from your speakers: a live stream, a video, an online radio station — through the sound card's loopback. Press Record to start and Stop to finish; a level meter shows the incoming signal.
Test tones and noise (Tools ▸ Tone Generator…).
The generator creates test signals, useful for calibrating a sound system, testing filters or creating reference material. Choose the signal type, then set the frequency, amplitude (in dB) and duration; for some types you can choose the stereo mode (Decorrelated, Phase Offset, Haas Delay).
| Available types |
|---|
| Sine · White Noise · Pink Noise · Sweep · IMD Test · IMD 19+20 kHz · Tone Burst Stereo · Random Burst Stereo · Square Wave · Saw Wave · Triangle Wave |
Speech synthesis and transcription (Voice&Text menu).
With Voice&Text ▸ Text-to-Speech Module ▸ System Voice… you generate spoken audio from text, using the synthesis voices installed in Windows. If the computer has no usable voices, the program prompts you to install one from the Windows language settings.
With Voice&Text ▸ Speech-to-Text Module ▸ Convert Speech to Text you transcribe the portion between the start and end markers into text. The result appears in the Transcript window, with word timings when available.
Models/en folder (English). The audio is converted to mono and resampled to 16 kHz before recognition. If the model is missing, the program reports that it was not found: it must be installed in the folder indicated. Since it works offline, transcription needs no connection, but it depends on the quality of the model and of the recording. Choosing the driver and device (Audio Device menu) and the default values (Settings menu).
From the Audio Device menu you select the output driver and device. From the Settings menu, instead, you set the default values: Export Settings… (MP3/WAV format, CBR/VBR mode, bitrate) and DSP Settings… (startup presets and chain order). The choice of driver affects latency and compatibility:
| Driver | Characteristics |
|---|---|
| WASAPI — Shared | Shared mode: coexists with other apps. It is the right choice if you also want to record system audio. |
| WASAPI — Exclusive | Exclusive access to the device: lower latency and full quality, but it blocks audio from other apps and prevents system recording. |
| ASIO | Very low-latency driver (professional sound cards). Always exclusive; requires an installed ASIO driver. |
The complete reference — keep it handy.
| Key | Action |
|---|---|
| Space | Play / Stop |
| + | Insert a cut marker (creates a region) |
| − | Remove the cut marker |
| Ctrl+Shift+X | Remove all cut markers |
| Key / gesture | Action |
|---|---|
| Mouse wheel | Zoom in / out |
| Shift + wheel | Scroll (pan) |
| Shift + drag | Horizontal “accordion” zoom |
| ↑ / ↓ | Zoom in / out |
| ← / → | Scroll left / right |
| O | Full view (zoom all the way out) |
| P | Center the view on the playhead |
| S | Center the view on the start marker |
| E | Center the view on the end marker |
| I | Move the playhead to the start marker |
| Key | Action |
|---|---|
| Ctrl+W | Insert Silence (silences the selection) |
| Ctrl+R | Reverse (reverses the selection) |
| Ctrl+P | Paste |
| Key | Action |
|---|---|
| F9 | Show / hide the DSP window |
| F10 | Show / hide the TIMELINE overlay (markers, regions and cuts) |
The technical terms you will come across in the program, in brief.
| Term | Meaning |
|---|---|
| LUFS | Unit of measurement of perceived loudness (EBU R128 standard). The yardstick streaming services use to match volumes. |
| dBFS | Decibels relative to digital full scale; 0 dBFS is the maximum before clipping. |
| True Peak (dBTP) | The true peak of the signal after conversion to analog: it can exceed the digital peaks and cause distortion. |
| Headroom | Margin between the signal level and 0 dBFS. |
| Threshold | Level above which a processor (compressor, de-esser) starts to act. |
| Ratio | Reduction ratio applied above the threshold (e.g. 4:1). |
| Attack / Release | How quickly an effect kicks in and releases. |
| Knee | How gradual (soft) or abrupt (hard) the onset of compression is around the threshold. |
| Q | Width of an EQ filter's action: high = narrow, low = broad. |
| Sibilants | The “s/sh/z” sounds, high-frequency energy that the de-esser softens. |
| Playhead | The cursor that shows the playback position. |
| Region | Portion of audio between two cut markers, with its own DSP chain. |
| DSP | Digital Signal Processing: the digital processing of sound (the effects in the chain). |
| CBR / VBR | Constant / variable bitrate in MP3. |
| WASAPI / ASIO | Windows audio driver systems; ASIO is low-latency, for pro sound cards. |
Quick answers to the most common situations.
In the Audio Device menu, check that the output device is the right one. If you chose WASAPI Exclusive or ASIO and another app is holding the device, switch to WASAPI Shared.
It is almost always WASAPI Exclusive mode: it prevents other apps from playing sound, so there is nothing to record. Set WASAPI Shared and try again.
The Vosk model is missing: it must be in the Models/en folder next to the program. Download the language model and place it there.
Install a voice from the Windows settings (Time & Language ▸ Speech). The program sees both SAPI5 and OneCore voices.
The gain applied is the smaller of Gain to Target and Gain Allowed: if raising the level all the way to the target would exceed the ceiling, the program stops short so as not to distort. To go higher, you first need to reduce the peaks (compressor / soft clipper).
Probably the Noise Reduction is too aggressive: lower Sensitivity and/or Reduction. Noise reduction should only be pushed until the background disappears, no further.
Check the Stereo Widener: delay-based widening can weaken the signal in mono. Reduce the mix or turn it off for export if the material has to hold up in mono.
CleanCutAudio user manual — reference document generated from the application's features. The numeric values (parameter ranges and factory presets) reflect the program's current configuration and may change in later versions.